[asterisk-dev] RTP- INVITE

Jared Smith jsmith at digium.com
Tue May 13 06:22:57 CDT 2008

On Tue, 2008-05-13 at 12:02 +0200, 0617260639 at alu.uma.es wrote:
> I want direct RTP traffic between the terminals but I've noticed that
> before that happens there is a short RTP flow between the terminals
> and Asterisk.

You can try setting "directrtpsetup=yes" in sip.conf, but please be
aware that it is labeled as experimental, and I've seen plenty of cases
where it doesn't work correctly. 

> Why is that?

Because Asterisk is a back-to-back user agent, and not a SIP proxy.  If
you learn the difference between a B2BUA and a proxy, you'll have a much
better understanding of how Asterisk fits in the world of telephony.

> I've also notices that there are several SIP INVITE messages during
> the RTP flow between terminal. Why does it send more than the initial

Again, because Asterisk is not a proxy.  If you *do* want a proxy (and
it sounds like you do), you might want to check out the OpenSER project.

Jared Smith
Training Manager
Digium, Inc.

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